carl Lougher wrote: > Howdy, > Was there ever a fix for this? > > I have Trix 2.6 running asterisk 1.4 and have to set an extension with > call-limit=1. However that user can no longer do attended transfers from > Linkys 962 ip phone. > > Is there anyway around this? > > Cheers, > Taff.. >
Yes, set call-limit to something else :P Seriously though, there's no "fix" for that since it is behaving exactly as it should. When attempting to transfer the call, Asterisk has no way of knowing that the new SIP INVITE it receives (in order to call the transfer target) is an attempt to transfer the call. It appears that the same SIP peer is attempting to make a second call. Since the call-limit is set to 1, Asterisk rejects the second call attempt. I haven't tried this yet, but it may actually be possible to use DTMF transfers when the call limit is that low since Asterisk is the one that actually initiates the new call to the transfer target instead of the transferer's phone. To use DTMF transfers, you need to set a DTMF sequence in features.conf and use the 't' or 'T' flag (depending on which party should have the ability to transfer the call) in your calls to Dial() or Queue(). Why do you have the call-limit set to 1, anyway? Mark Michelson _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
